April 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSWITCH Weekly News and Notes
by Michael Collins 29 Apr '13

29 Apr '13

Hello Friends, We are looking forward to a busy but good week of activity. Last week was busy as well. On last week's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2013_04_24>we had an engineer from JiraSoft <http://www.jerasoft.net/products/vcs/index.html> join us to answer some questions about the newly released VCS 2.4.3. We are happy to have this new version as part of the FreeSWITCH ecosystem<http://freeswitch.org/node/447>. This week <http://wiki.freeswitch.org/wiki/FS_weekly_2013_05_01> we welcome Omar from OrecX <http://www.orecx.com/>. OrecX creates commercial and open source telephony recording solutions. We look forward to learning more about how we can use OrecX with FreeSWITCH. One other bit of good news: All of the content for the new FreeSWITCH book<http://www.packtpub.com/freeswitch-1-2/book>has been submitted to the publishers. We are doing a bit of editing on one of the chapters but otherwise we are basically done. We anticipate a June release. Take care and have another fantastic week! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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ERROR 404 BAD HEADER
by pratham2013 29 Apr '13

29 Apr '13

Hello, I'm facing very strange issue. I has setup extensions in FreeSwitch and I can login and make calls but on of my customer can not register with same extension and they are getting error 400 bad contact header so I could not understand how to solve this issue. Please anyone can help? Thanks Manish Khilwani -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/ERROR-404-BAD-HEADER-tp759004… Sent from the freeswitch-users mailing list archive at Nabble.com.

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BAD CONTACT HEADER
by pratham2013 29 Apr '13

29 Apr '13

Hello,My SIP device is giving bad contact header when I'm trying to register via softphone. Below are the SIP TRACE I have posted. freeswitch@domU-12-31-39-02-85-BD> recv 569 bytes from udp/[189.234.223.22]:10604 at 19:32:38.981695: ------------------------------------------------------------------------ REGISTER sip:54.225.79.159;transport=udp SIP/2.0 Via: SIP/2.0/UDP 189.234.223.22:10604;branch=z9hG4bK-d8754z-7ac384e9b8979fec-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1000@189.234.223.22:10604;rinstance=830f25b897d4ecb9;transport=udp> To: "1000"<sip:1000@54.225.79.159> From: "1000"<sip:1000@54.225.79.159>;tag=a4771acf Call-ID: Nzk3Mjk3MmE0YjY5MjkxNmNiZGQyMDdmODI1NjhmNWM CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 4.5 stamp 69607 Content-Length: 0 ------------------------------------------------------------------------ send 672 bytes to udp/[189.234.223.22]:10604 at 19:32:38.982923: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 189.234.223.22:10604;branch=z9hG4bK-d8754z-7ac384e9b8979fec-1---d8754z-;rport=10604 From: "1000"<sip:1000@54.225.79.159>;tag=a4771acf To: "1000" <sip:1000@54.225.79.159>;tag=120rgNHa202Xc Call-ID: Nzk3Mjk3MmE0YjY5MjkxNmNiZGQyMDdmODI1NjhmNWM CSeq: 1 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.5.1b+git~20130406T213201Z~1e251568cc Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: precondition, path, replaces WWW-Authenticate: Digest realm="54.225.79.159", nonce="8b341bb6-0b60-45bd-9c12-d2fba508c185", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ 2013-04-25 19:32:38.982782 [WARNING] sofia_reg.c:1587 SIP auth challenge (REGISTER) on sofia profile 'internal' for [1000(a)54.225.79.159] from ip 189.234.223.22 recv 834 bytes from udp/[189.234.223.22]:10604 at 19:32:39.159706: ------------------------------------------------------------------------ REGISTER sip:54.225.79.159;transport=udp SIP/2.0 Via: SIP/2.0/UDP 189.234.223.22:10604;branch=z9hG4bK-d8754z-5015538769fafb1a-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1000@189.234.223.22:10604;rinstance=830f25b897d4ecb9;transport=udp> To: "1000"<sip:1000@54.225.79.159> From: "1000"<sip:1000@54.225.79.159>;tag=a4771acf Call-ID: Nzk3Mjk3MmE0YjY5MjkxNmNiZGQyMDdmODI1NjhmNWM CSeq: 2 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 4.5 stamp 69607 Authorization: Digest username="1000",realm="54.225.79.159",nonce="8b341bb6-0b60-45bd-9c12-d2fba508c185",uri="sip:54.225.79.159;transport=udp",response="428a6156713f632381221a14fbd4fd52",cnonce="d7aa657b759663a6b51d81c568efdbdc",nc=00000001,qop=auth,algorithm=MD5 Content-Length: 0 ------------------------------------------------------------------------ send 674 bytes to udp/[189.234.223.22]:10604 at 19:32:39.162185: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 189.234.223.22:10604;branch=z9hG4bK-d8754z-5015538769fafb1a-1---d8754z-;rport=10604 From: "1000"<sip:1000@54.225.79.159>;tag=a4771acf To: "1000" <sip:1000@54.225.79.159>;tag=2BtHjg2DZ9rgr Call-ID: Nzk3Mjk3MmE0YjY5MjkxNmNiZGQyMDdmODI1NjhmNWM CSeq: 2 REGISTER Contact: <sip:1000@189.234.223.22:10604;rinstance=830f25b897d4ecb9;transport=udp>;expires=3600 Date: Thu, 25 Apr 2013 19:32:39 GMT User-Agent: FreeSWITCH-mod_sofia/1.5.1b+git~20130406T213201Z~1e251568cc Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ recv 829 bytes from udp/[189.234.223.22]:10604 at 19:32:39.339465: ------------------------------------------------------------------------ REGISTER sip:54.225.79.159;transport=udp SIP/2.0 Via: SIP/2.0/UDP 189.234.223.22:10604;branch=z9hG4bK-d8754z-b7e8bd30145a2cb4-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1000@189.234.223.22:10604;rinstance=830f25b897d4ecb9;transport=udp>;expires=0 To: "1000"<sip:1000@54.225.79.159> From: "1000"<sip:1000@54.225.79.159>;tag=a4771acf Call-ID: Nzk3Mjk3MmE0YjY5MjkxNmNiZGQyMDdmODI1NjhmNWM CSeq: 3 REGISTER Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 4.5 stamp 69607 Authorization: Digest username="1000",realm="54.225.79.159",nonce="8b341bb6-0b60-45bd-9c12-d2fba508c185",uri="sip:54.225.79.159;transport=udp",response="a957575e31cdba10c7be41a9db26eda2",cnonce="70b4da67f9ae2f1ab6f2599658e00548",nc=00000002,qop=auth,algorithm=MD5 Content-Length: 0 ------------------------------------------------------------------------ send 578 bytes to udp/[189.234.223.22]:10604 at 19:32:39.341224: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 189.234.223.22:10604;branch=z9hG4bK-d8754z-b7e8bd30145a2cb4-1---d8754z-;rport=10604 From: "1000"<sip:1000@54.225.79.159>;tag=a4771acf To: "1000" <sip:1000@54.225.79.159>;tag=3mKamBKHvjF3K Call-ID: Nzk3Mjk3MmE0YjY5MjkxNmNiZGQyMDdmODI1NjhmNWM CSeq: 3 REGISTER Date: Thu, 25 Apr 2013 19:32:39 GMT User-Agent: FreeSWITCH-mod_sofia/1.5.1b+git~20130406T213201Z~1e251568cc Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ recv 840 bytes from udp/[189.234.223.22]:10604 at 19:32:39.517716: ------------------------------------------------------------------------ REGISTER sip:54.225.79.159;transport=udp SIP/2.0 Via: SIP/2.0/UDP 189.234.223.22:10604;branch=z9hG4bK-d8754z-c43ad0c3c11ccbe2-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1000@189.234.223.22:10604:10604;transport=udp;rinstance=6860708eab0fb92b> To: "1000"<sip:1000@54.225.79.159> From: "1000"<sip:1000@54.225.79.159>;tag=a4771acf Call-ID: Nzk3Mjk3MmE0YjY5MjkxNmNiZGQyMDdmODI1NjhmNWM CSeq: 4 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 4.5 stamp 69607 Authorization: Digest username="1000",realm="54.225.79.159",nonce="8b341bb6-0b60-45bd-9c12-d2fba508c185",uri="sip:54.225.79.159;transport=udp",response="89e4f1175f553d3fcc27811f956dcf1a",cnonce="418a6e48bdec6d7e8c4ae68f1e80dd27",nc=00000003,qop=auth,algorithm=MD5 Content-Length: 0 ------------------------------------------------------------------------ send 334 bytes to udp/[189.234.223.22]:10604 at 19:32:39.517869: ------------------------------------------------------------------------ SIP/2.0 400 Bad Contact Header Via: SIP/2.0/UDP 189.234.223.22:10604;branch=z9hG4bK-d8754z-c43ad0c3c11ccbe2-1---d8754z-;rport=10604 From: "1000" <sip:1000@54.225.79.159>;tag=a4771acf To: "1000" <sip:1000@54.225.79.159>;tag=4Xc3N63mSU5NF Call-ID: Nzk3Mjk3MmE0YjY5MjkxNmNiZGQyMDdmODI1NjhmNWM CSeq: 4 REGISTER Content-Length: 0 ------------------------------------------------------------------------ At last its sending contact like this : Contact: <sip:1000@189.234.223.22:10604:10604;transport=udp;rinstance=6860708eab0fb92b>Port 10604 is attached 2 times and I think this may be the issue. I have also turned off SIP ALG in router.Please help me as its very critical issue.ThanksManish Khilwani -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/BAD-CONTACT-HEADER-tp7590229.… Sent from the freeswitch-users mailing list archive at Nabble.com.

2 1

problem with "tel:" uri in "To:" header
by Mickael Hubert 29 Apr '13

29 Apr '13

Hi list, I have an problem with my tests. IPBX: 10.84.8.100 --> Opensips 10.84.8.20 --> freeswitch 10.84.8.6 --> SIP provider I'm testing tel URI in my freeswitch, and all it's OK, but when I testing in my INVITE: To:<tel:+33XXXXXXX@10.84.8.20> in all responses (183,200) come from my freeswith, I have a "bug" in my contact header with: Contact: <sip:+33XXXXXXX@*10.84.8.20@10.84.8.6:5060*;transport=udp> It originates from the To header with "tel:" beacause when I do the test with "normal" "sip:" header, all it's OK in my 183,200 reponses. How can I solve this problem please ? thanks in advance -- Cordialement Hubert Mickaël Ingénieur VOIP - Hexanet --

1 2

video call to h323 soft phone via freeswitch from sip soft phone
by Navnath Sonavne 29 Apr '13

29 Apr '13

Hi All, My video call is failing from X-Lite softphone(SIP protocol) to Ekiga softphone(h323 protocol). Ekiga softphone is answering the call but there is NO VIDEO transfer only audio transfer. My X-Lite softphone is at 192.168.9.165 and Ekiga softphone is at 192.168.8.27. Freeswitch server ip : 192.168.8.41 I have made changes in Local_Extension of default.xml such that if any registered extension has been dialed ,call will be forwarded to softphones at 192.168.8.27. <extension name="Local_Extension"> <condition field="destination_number" expression="^(\d{4})$"> <action application="export" data="dialed_extension=$1"/> <action application="set" data="effective_caller_id_number=${dialed_extension}"/> <action application="set" data="PEER_IP=192.168.8.103"/> <action application="set" data="call_timeout=30"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="proxy_media=true"/> <action application="bridge" data="h323/$1@${PEER_IP}"/> </condition> </extension>. Here is my freeswitch console log for video call from X-Lite softphone(SIP protocol) to Ekiga softphone(h323 protocol): 2013-04-29 02:34:00.220733 [NOTICE] switch_channel.c:978 New Channel sofia/internal/1100(a)192.168.8.41 [c7188b62-81ca-4225-9c8e-ccdb32e97af2] 2013-04-29 02:34:00.260553 [INFO] mod_dialplan_xml.c:557 Processing navnath <1100>->1101 in context default 2013-04-29 02:34:00.479931 [NOTICE] switch_channel.c:978 New Channel h323/1101(a)192.168.8.27 [cf6f458c-c443-40ea-ab5a-6660853b5df8] 2013-04-29 02:34:00.479931 [WARNING] switch_core_session.c:635 h323/1101(a)192.168.8.27 does not support the proxy feature, disabling. 2013-04-29 02:34:00.479931 [WARNING] switch_core_session.c:648 h323/1101(a)192.168.8.27 does not support the proxy feature, disabling. 2013-04-29 02:34:00.479931 [NOTICE] mod_h323.cpp:856 capability 'G.711-ALaw-64k' format 'PCMA' 8 2013-04-29 02:34:00.479931 [NOTICE] mod_h323.cpp:856 capability 'G.711-uLaw-64k' format 'PCMU' 0 2013-04-29 02:34:00.479931 [NOTICE] mod_h323.cpp:856 capability 'GSM-06.10' format 'GSM' 3 2013-04-29 02:34:00.479931 [NOTICE] mod_h323.cpp:856 capability 'G.729A/B' format 'G729' 18 2013-04-29 02:34:00.479931 [NOTICE] mod_h323.cpp:856 capability 'G.729A' format 'G729' 18 2013-04-29 02:34:00.479931 [NOTICE] mod_h323.cpp:856 capability 'G.729B' format 'G729b' 18 2013-04-29 02:34:00.479931 [NOTICE] mod_h323.cpp:856 capability 'G.729' format 'G729' 18 (.navn 61 74 68 00 6c 05 81 31 31 30 31 70 05 81 31 31 ath.l..1101p..11 30 31 7e 03 06 05 20 b8 06 00 08 91 4a 00 06 01 01~... .....J... 01 80 44 34 28 c0 09 00 00 3d 14 46 72 65 65 53 ..D4(....=.FreeS 57 49 54 4 2013-04-29 02:34:00.600359 [NOTICE] mod_h323.cpp:1147 Ring-Ready h323/1101(a)192.168.8.27! 2013-04-29 02:34:00.600359 [INFO] h323ep.cxx:3048 Received alerting PDU. (.. 2013-04-29 02:34:00.620246 [NOTICE] mod_sofia.c:2035 Ring-Ready sofia/internal/1100(a)192.168.8.41! 2013-04-29 02:34:00.620246 [NOTICE] switch_ivr_originate.c:521 Ring Ready sofia/internal/1100(a)192.168.8.41! 2013-04-29 02:34:12.819989 [NOTICE] mod_h323.cpp:1012 Hangup h323/1101(a)192.168.8.27 [CS_CONSUME_MEDIA] [CALL_REJECTED] 2013-04-29 02:34:12.839928 [INFO] mod_dptools.c:3088 Originate Failed. Cause: CALL_REJECTED 2013-04-29 02:34:12.839928 [NOTICE] mod_dptools.c:3208 Hangup sofia/internal/1100(a)192.168.8.41 [CS_EXECUTE] [CALL_REJECTED] 2013-04-29 02:34:12.839928 [INFO] h323.cxx:1273 Connection ip$localhost/18648 terminated. 2013-04-29 02:34:12.839928 [NOTICE] switch_core_session.c:1566 Session 1 (sofia/internal/1100(a)192.168.8.41) Ended 2013-04-29 02:34:12.839928 [NOTICE] switch_core_session.c:1570 Close Channel sofia/internal/1100(a)192.168.8.41 [CS_DESTROY] 2013-04-29 02:34:13.819657 [NOTICE] switch_core_session.c:1566 Session 2 (h323/1101(a)192.168.8.27) Ended 2013-04-29 02:34:13.819657 [NOTICE] switch_core_session.c:1570 Close Channel h323/1101(a)192.168.8.27 [CS_DESTROY] 2013-04-29 02:35:51.759989 [NOTICE] switch_channel.c:978 New Channel sofia/internal/1100(a)192.168.8.41 [23720e51-1818-4a08-aa36-958ebc19cf45] 2013-04-29 02:35:51.799989 [INFO] mod_dialplan_xml.c:557 Processing navnath <1100>->1101 in context default 2013-04-29 02:35:52.019984 [NOTICE] switch_channel.c:978 New Channel h323/1101(a)192.168.8.27 [09ee14e2-dffe-435d-b58f-6db0b8ac1314] 2013-04-29 02:35:52.019984 [WARNING] switch_core_session.c:635 h323/1101(a)192.168.8.27 does not support the proxy feature, disabling. 2013-04-29 02:35:52.019984 [WARNING] switch_core_session.c:648 h323/1101(a)192.168.8.27 does not support the proxy feature, disabling. 2013-04-29 02:35:52.019984 [NOTICE] mod_h323.cpp:856 capability 'G.711-ALaw-64k' format 'PCMA' 8 2013-04-29 02:35:52.019984 [NOTICE] mod_h323.cpp:856 capability 'G.711-uLaw-64k' format 'PCMU' 0 2013-04-29 02:35:52.019984 [NOTICE] mod_h323.cpp:856 capability 'GSM-06.10' format 'GSM' 3 2013-04-29 02:35:52.019984 [NOTICE] mod_h323.cpp:856 capability 'G.729A/B' format 'G729' 18 2013-04-29 02:35:52.019984 [NOTICE] mod_h323.cpp:856 capability 'G.729A' format 'G729' 18 2013-04-29 02:35:52.019984 [NOTICE] mod_h323.cpp:856 capability 'G.729B' format 'G729b' 18 2013-04-29 02:35:52.019984 [NOTICE] mod_h323.cpp:856 capability 'G.729' format 'G729' 18 (.navn 61 74 68 00 6c 05 81 31 31 30 31 70 05 81 31 31 ath.l..1101p..11 30 31 7e 03 06 05 20 b8 06 00 08 91 4a 00 06 01 01~... .....J... 01 80 44 34 28 c0 09 00 00 3d 14 46 72 65 65 53 ..D4(....=.FreeS 57 49 54 4 2013-04-29 02:35:52.799941 [NOTICE] mod_h323.cpp:1147 Ring-Ready h323/1101(a)192.168.8.27! 2013-04-29 02:35:52.799941 [INFO] h323ep.cxx:3048 Received alerting PDU. (.. 2013-04-29 02:35:52.819889 [NOTICE] mod_sofia.c:2035 Ring-Ready sofia/internal/1100(a)192.168.8.41! 2013-04-29 02:35:52.819889 [NOTICE] switch_ivr_originate.c:521 Ring Ready sofia/internal/1100(a)192.168.8.41! 2013-04-29 02:36:06.719989 [INFO] h323ep.cxx:3115 Received connect PDU. 2013-04-29 02:36:06.719989 [NOTICE] mod_h323.cpp:1258 Channel [h323/1101(a)192.168.8.27] has been answered 2013-04-29 02:36:06.739922 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1100(a)192.168.8.41! 2013-04-29 02:36:06.739922 [NOTICE] switch_ivr_originate.c:3411 Channel [sofia/internal/1100(a)192.168.8.41] has been answered 2013-04-29 02:36:12.959989 [NOTICE] mod_h323.cpp:1012 Hangup h323/1101(a)192.168.8.27 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2013-04-29 02:36:12.959989 [INFO] h323.cxx:1273 Connection ip$localhost/18649 terminated. 2013-04-29 02:36:12.979942 [NOTICE] switch_ivr_bridge.c:1516 Hangup sofia/internal/1100(a)192.168.8.41 [CS_EXECUTE] [NORMAL_CLEARING] 2013-04-29 02:36:12.979942 [NOTICE] switch_core_session.c:1566 Session 3 (sofia/internal/1100(a)192.168.8.41) Ended 2013-04-29 02:36:12.979942 [NOTICE] switch_core_session.c:1570 Close Channel sofia/internal/1100(a)192.168.8.41 [CS_DESTROY] 2013-04-29 02:36:13.979874 [NOTICE] switch_core_session.c:1566 Session 4 (h323/1101(a)192.168.8.27) Ended 2013-04-29 02:36:13.979874 [NOTICE] switch_core_session.c:1570 Close Channel h323/1101(a)192.168.8.27 [CS_DESTROY] -Navnath.

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http://wiki.freeswitch.org/wiki/Java_ESL_Client, how I can make a connection before starting to handle Outbound connection ?
by Marwan Idriss 29 Apr '13

29 Apr '13

Greeting, bellow is an example to handle outbound connection and make hung up: but it never show How to make the connection in first place is their an example in main of how this function work ? /* * Copyright 2010 david varnes. * * Licensed under the Apache License, version 2.0 (the "License"); * you may not use this file except in compliance with the License. * You may obtain a copy of the License at: * * http://www.apache.org/licenses/LICENSE-2.0 * * Unless required by applicable law or agreed to in writing, software * distributed under the License is distributed on an "AS IS" BASIS, * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. * See the License for the specific language governing permissions and * limitations under the License. */ package org.freeswitch.esl.client.outbound.example; import org.freeswitch.esl.client.outbound.AbstractOutboundClientHandler; import org.freeswitch.esl.client.transport.SendMsg; import org.freeswitch.esl.client.transport.event.EslEvent; import org.freeswitch.esl.client.transport.message.EslMessage; import org.freeswitch.esl.client.transport.message.EslHeaders.Name; import org.jboss.netty.channel.Channel; import org.jboss.netty.channel.ChannelHandlerContext; /** * Simple example of a handler for outbound connection from FreeSWITCH server. * This class will log some of the FreeSWTICH call channel variables and * then hangup the call. * * @author david varnes */ public class SimpleHangupOutboundHandler extends AbstractOutboundClientHandler { @Override protected void handleConnectResponse( ChannelHandlerContext ctx, EslEvent event ) { log.info( "Received connect response [{}]", event ); if ( event.getEventName().equalsIgnoreCase( "CHANNEL_DATA" ) ) { // this is the response to the initial connect log.info( "======================= incoming channel data =============================" ); log.info( "Event-Date-Local: [{}]", event.getEventDateLocal() ); log.info( "Unique-ID: [{}]", event.getEventHeaders().get( "Unique-ID" ) ); log.info( "Channel-ANI: [{}]", event.getEventHeaders().get( "Channel-ANI" ) ); log.info( "Answer-State: [{}]", event.getEventHeaders().get( "Answer-State" ) ); log.info( "Caller-Destination-Number: [{}]", event.getEventHeaders().get( "Caller-Destination-Number" ) ); log.info( "======================= = = = = = = = = = = = =============================" ); // now hangup the call hangupCall( ctx.getChannel() ); } else { throw new IllegalStateException( "Unexpected event after connect: [" + event.getEventName() + ']' ); } } @Override protected void handleEslEvent( ChannelHandlerContext ctx, EslEvent event ) { log.info( "Received event [{}]", event ); } private void hangupCall( Channel channel ) { SendMsg hangupMsg = new SendMsg(); hangupMsg.addCallCommand( "execute" ); hangupMsg.addExecuteAppName( "hangup" ); EslMessage response = sendSyncMultiLineCommand( channel, hangupMsg.getMsgLines() ); if ( response.getHeaderValue( Name.REPLY_TEXT ).startsWith( "+OK" ) ) { log.info( "Call hangup successful" ); } else { log.error( "Call hangup failed: [{}}", response.getHeaderValue( Name.REPLY_TEXT ) ); } } }

3 2

Re: [Freeswitch-users] Freeswitch silent crash
by Mr Nathan Downes 28 Apr '13

28 Apr '13

Hi, Whats the Jira for this? I am having the same problem and use a lot of lua scripts.. only started happening when I upgraded to latest stable. It's production so in trouble :( Thanks, Nathan -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of covici(a)ccs.covici.com Sent: Friday, 26 April 2013 4:43 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Freeswitch silent crash echo 1 >/proc/sys/fs/suid_dumpable for any process like that -- I got bit by that for quite a while. Ken Rice <krice(a)freeswitch.org> wrote: > It should, > > If you are running as a non-root user on linux there is a sysctl that > should be referenced on the wiki (I can never remember it off the top > of my head) you will need to set properly for the non-root user to dump the core file. > > Look at the sysctl kernel.core_pattern for the pattern of the file > name used. > > K > > On 4/25/13 9:43 AM, "Antoine Reversat" <antoine(a)fibrenoire.ca> wrote: > > > Ok so this would have dumped a core if I had set up my environnment properly ? > > > > > > > > > > On Thu, Apr 25, 2013 at 10:40 AM, Ken Rice <krice(a)freeswitch.org> wrote: > >> You should have core dumping enable for freeswitch, then get a backtrace... > >> > >> See http://wiki.freeswitch.org/wiki/Debugging_Freeswitch > >> > >> > >> > >> On 4/25/13 9:23 AM, "Antoine Reversat" <antoine(a)fibrenoire.ca > >> <http://antoine@fibrenoire.ca> > wrote: > >> > >>> Doing what you told me yielded an interesting result. > >>> > >>> Freeswitch crashed with this : > >>> > >>> freeswitch: pthread_mutex_lock.c:62: __pthread_mutex_lock: > >>> Assertion `mutex->__data.__owner == 0' failed. > >>> > >>> Should I run freeswitch in gdb to get why this happens ? > >>> > >>> > >>> On Tue, Apr 23, 2013 at 11:16 AM, Anthony Minessale > >>> <anthony.minessale(a)gmail.com <http://anthony.minessale@gmail.com> > wrote: > >>>> See if maybe one of your scripts is calling the exit syscall. > >>>> This can terminate the whole process. > >>>> Run FS in a unix screen session and run it from the console so > >>>> you can see what happens in the stdout. > >>>> > >>>> > >>>> > >>>> On Tue, Apr 23, 2013 at 9:17 AM, Antoine Reversat > >>>> <antoine(a)fibrenoire.ca <http://antoine@fibrenoire.ca> > wrote: > >>>>> Hi, > >>>>> > >>>>> I have a weird issue where once in a while (apparently > >>>>> randomly) freeswitch crashes silently (that is I don't see any > >>>>> mention of freeswitch having sgfaulted in the output of dmesg > >>>>> and the logs aren't helpgul either). > >>>>> > >>>>> The version of freeswitch is the latest 1.2 stable branch from > >>>>> git packaged by myself for ubuntu. > >>>>> > >>>>> The OS is Ubuntu 12.04 with the latest 3.2.0-39-generic > >>>>> #62-Ubuntu kernel. > >>>>> > >>>>> I have 2 servers and both randomly crash in the same way so I > >>>>> ruled out a hardware issue. > >>>>> > >>>>> Our whole call processing is done in LUA but the crash occurs > >>>>> with no call in progress. > >>>>> > >>>>> I know it's not a lot to work with but if you have pointers on > >>>>> how to get more information about what happens I'd be happy to try it out. > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #freeswitch > > > ---------------------------------------------------- > Alternatives: > > ---------------------------------------------------- > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici(a)ccs.covici.com _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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RTCP
by Alex Lake 26 Apr '13

26 Apr '13

Interesting to see a reference to RTCP in a recent thread. I looked it up on the wiki and there's not much there about how to use it for giving diagnostic output on actual calls. I would be interested in investigating (and would be happy to update the Wiki) but where should I start?!

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How to set Callee ID Name globally or on gateway level
by Deon Vermeulen 26 Apr '13

26 Apr '13

Hi I've stumbled upon the following links: http://freeswitch-users.2379917.n2.nabble.com/effective-callee-id-name-beha… http://lists.freeswitch.org/pipermail/freeswitch-dev/2011-November/005355.h… Am I correct to say that these set the callee_id_name to be passed in the SIP INVITE when configured on the extension? We require the following INVITE format to be sent over a specific gateway: INVITE sip:1234567890@192.168.200.153;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.200.130:5070;branch=z9hG4bK75b28f842f76d9c9e299a80c0;rport;X-DispCookie=1013;X-HwTRC=259446;X-DptMsg=1405 Route: <sip:192.168.200.153:5060;transport=udp;lr> Call-ID: bb4bcb34ec7041237fbfc1e63f687744(a)10.18.5.64 From: "0987654321"<sip:0987654321@192.168.200.130;transport=udp;user=phone>;tag=fd64e901-CC-1013-TRC-259446 To:"1234567890"<sip:1234567890@192.168.200.153;transport=udp;user=phone> How could one go about to achieve the above? Thanks for any help. Kind Regards Deon Vermeulen

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Friday Free For All Activate!
by Ken Rice 26 Apr '13

26 Apr '13

Hey Guys, Its that time of the week again... Lets get the Productivity Killer started... sip:888@conference.freeswitch.org Or if you need alternative access http://wiki.freeswitch.org/wiki/Weekly_Conference_Call_Calling_Instructions Join us... Todays topic is... Well its a free for all -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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